• The Journal of Korean Association of Computer Education
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• v.8 no.6
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• pp.75-83
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• 2005

In this paper we studied English listening and speaking test part of foreign language using web and VoiceXML-based education testing system, which is irrespective of time and space. The testing system interface based on VoiceXML consists of user registration module, testing module, and testing result module. User registration module registers user's name and ID, password in user database, and when a tester calls for testing, the User listens to the telephone sound supported by vxml scenario. After that, if a tester logs in, the tester is verified, In the VoiceXML-based education testing system, the manager can reduce time and effort for gaining testing result. The tester listens to the voice by scenario supported by VoiceXML markup language using wire/wireless telephone at any time or anywhere and can improve the effect of foreign language studying by valuating in voice directly. verified. In the VoiceXML-based education testing system, the manager can reduce time and effort for gaining testing result. The tester listens to the voice by scenario supported by VoiceXML markup language using wire/wireless telephone at any time or anywhere and can improve the effect of foreign language studying by valuating in voice directly.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.42-47
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• 2000

This paper describes a real-time implementation of a GSM-EFR (Global System for Mobil communications Enhanced Full Rate) speech coder using OakDSP core; a 16bit fixed-point Digital Signal Processor (DSP) by DSP Group, Inc. The real-time implemented speech coder required about 24MIPS for computation and 7.06K words and 12.19K words for code and data memory, respectively. The implemented GSM-EFR speech coder passes all of test vectors provided by ETSI (European Telecommunication Standard Institute), and perceptual speech quality measurement using MNB algorithm shows that the quality of the GSM-EFR speech coder is similar to the one of 32kbps ADPCM. The real-time implemented GSM-EFR speech coder which is the highest bit-rate mode of the GSM-AMR speech coder will be used as the basic structure of the GSM-AMR speech coder which is embedded in MODEM ASIC of IMT2000 asynchronous mode mobile station.

• The Journal of the Acoustical Society of Korea
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• v.23 no.5
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• pp.400-409
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• 2004

In this paper. a voice personality transformation method is proposed. which makes one person's voice sound like another person's voice. To transform the voice personality. vocal tract transfer function is used as a transformation parameter. Comparing with previous methods. the proposed method makes transformed speech closer to target speaker's voice in both subjective and objective points of view. Conversion between vocal tract transfer functions is implemented by classification of entire vector space followed by linear transformation for each cluster. LPC cepstrum is used as a feature parameter. A joint classification and transformation method is proposed, where optimum clusters and transformation matrices are simultaneously estimated in the sense of a minimum mean square error criterion. To evaluate the performance of the proposed method. transformation rules are generated from 150 sentences uttered by three male and on female speakers. These rules are then applied to another 150 sentences uttered by the same speakers. and objective evaluation and subjective listening tests are performed.

• The KIPS Transactions:PartB
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• v.10B no.2
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• pp.213-218
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• 2003

In this paper, we implement a crossword game, which operate by speech. The CAA (Cross Array Algorithm) produces the crossword array randomly and automatically using an domain-dictionary. For producing the crossword array, we construct seven domain-dictionaries. The crossword game is operated by a mouse and a keyboard and is also operated by speech. For the user interface by speech, we use a speech recognizer and a speech synthesizer and this provide more comfortable interface to the user. The efficiency evaluation of CAA is performed by estimating the processing times of producing the crossword array and the generation ratio of the crossword array. As the results of the CAA's efficiency evaluation, the processing times is about 10ms and the generation ratio of the crossword array is about 50%. Also, the recognition rates were 95.5%, 97.6% and 96.2% for the window sizes of "$7{\times}7$", "$9{\times}9$," and "$11{\times}11$" respectively.}11$" respectively.vely.

• The Journal of the Acoustical Society of Korea
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• v.26 no.8
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• pp.390-396
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• 2007

In this letter, we propose a novel approach to improve the performance of speech/music classification for the selectable mode vocoder(SMV) of 3GPP2 using the Gaussian mixture model(GMM) which is based on the expectation-maximization(EM) algorithm. We first present an effective analysis of the features and the classification method adopted in the conventional SMV. And then feature vectors which are applied to the GMM are selected from relevant Parameters of the SMV for the efficient speech/music classification. The performance of the proposed algorithm is evaluated under various conditions and yields better results compared with the conventional scheme of the SMV.

• The Journal of the Acoustical Society of Korea
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• v.26 no.5
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• pp.207-213
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• 2007

In this research, we propose an efficient Speech/Music discrimination method that uses spectrum analysis and neural network. The proposed method extracts the duration feature parameter(MSDF) from a spectral peak track by analyzing the spectrum, and it was used as a feature for Speech/Music discriminator combined with the MFSC. The neural network was used as a Speech/Music discriminator, and we have reformed various experiments to evaluate the proposed method according to the training pattern selection, size and neural network architecture. From the results of Speech/Music discrimination, we found performance improvement and stability according to the training pattern selection and model composition in comparison to previous method. The MSDF and MFSC are used as a feature parameter which is over 50 seconds of training pattern, a discrimination rate of 94.97% for speech and 92.38% for music. Finally, we have achieved performance improvement 1.25% for speech and 1.69% for music compares to the use of MFSC.

• The Journal of the Acoustical Society of Korea
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• v.27 no.5
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• pp.256-261
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• 2008

In this paper, we propose a novel approach to improve the performance of minima controlled recursive averaging (MCRA) which is based on the conditional maximum a posteriori criterion. A crucial component of a practical speech enhancement system is the estimation of the noise power spectrum. One state-of-the-art approach is the minima controlled recursive averaging (MCRA) technique. The noise estimate in the MCRA technique is obtained by averaging past spectral power values based on a smoothing parameter that is adjusted by the signal presence probability in frequency subbands. We improve the MCRA using the speech presence probability which is the a posteriori probability conditioned on both the current observation the speech presence or absence of the previous frame. With the performance criteria of the ITU-T P.862 perceptual evaluation of speech quality (PESQ) and subjective evaluation of speech quality, we show that the proposed algorithm yields better results compared to the conventional MCRA-based scheme.

• The Journal of the Acoustical Society of Korea
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• v.27 no.6
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• pp.319-324
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• 2008

In this paper, we propose a novel approach to improve the performance of speech/music classification for the selectable mode vocoder (SMV) of 3GPP2 using the discriminative weight training which is based on the minimum classification error (MCE) algorithm. We first present an effective analysis of the features and the classification method adopted in the conventional SMV. And then proposed the speech/music decision rule is expressed as the geometric mean of optimally weighted features which are selected from the SMV. The performance of the proposed algorithm is evaluated under various conditions and yields better results compared with the conventional scheme of the SMV.

• KAPE Magazine
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• s.258
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• pp.7-10
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• 2018

뭔가 한없이 분주하면서도 익숙한 흐름이 안정적인 느낌을 주는 오전의 농협음성축산물공판장 평가사 대기실. 눈에 띄는 한 무리가 보인다. 빠르게 오가는 사람들 속에서 누구보다 격하게 인사를 하는 모습에 옅은 긴장감과 각오가 묻어난다.

• Journal of Digital Contents Society
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• v.19 no.4
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• pp.621-630
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• 2018

This paper presents a case study on foreign language learning applications based on the speech recognition technology, aiming to grasp their current status and limitations of the technology applied to the foreign language speaking education, especially for English and French. As a result of examining the characteristics of the selected English and French applications by drawing on speech learning, it is shown that the use of speech recognition technology has the advantage of creating a speaking practice environment and giving feedback. However, in the case of feedback, there is a lack of appropriate calibration feedback which can help learners correct errors by themselves.